Hello Everyone Has anyone here every tried this successfully? I'm trying to stream CD quality audio through an I2S interface directly to a Delta Sigma DAC, and out to the headphone jack. I can hear the audio through the head phones connected to the jack on the board but the audio sounds HORRIBLE. I've been told that I need to implement an interpolation filter in between the delta sigma dac and the I2S receiver and that I can't send the audio directly to the DAC? I noticed that there was a wave file project given away with the Pipistrello board, so I'm assuming that someone has tried this and gotten it to work? So here is my set up FPGA: Spartan 6 lx45 Pipistrello board External Bluetooth Chip Connected via an I2S interface CD Quality Audio Streaming at 44.1kHz 24bit data cropped to 16 bits ( LSB cropped ) 16bit data feed to the Delta Sigma DAC running at 75Mhz So, I need an interpolation filter, or so I'm told, so do we have any laying around here? Any other suggestions or solutions to this problem? Thanks in advance for all of your help.

Hello. I'm working on a MAX5556 Audio DAC Wing and I thought I should share it here . It (obviously) has a MAX5556 Stereo DAC at its core, with an I2S-compatible interface and up to 50KHz sampling rate (perfect for standard 44.1 and 48 KHz sampling rates) and a resolution of 16 and 24 bits. The board does not contain any output buffer as I wanted the board to be as small as possible. It only has the MAX5556, a couple of passives and a 3.5mm stereo jack. I'll leave you with the schematics and a render of the board. Any criticism is well accepted I'm also prototyping a Megawing with an ADC, a DAC and an Ethernet controller to stream audio over ethernet. Schematic.pdf

While going through hamster's Intro to FPGA & VHDL book (great resource BTW), I had a go at outputting 8-bit audio at 11 KHz through a sigma-delta DAC. It worked, but the audio quality was pretty poor. Yes, 8-bit @ 11 KHz audio is low-fi, but the same audio data is much less noisy when played through a computer's sound-card. The DAC itself was clocked at 32 MHz, which I think should be more than enough for decent quality output. I set myself the goal of improving quality to be similar to a sound-card (without increasing the number of bits or the input sample-rate). So far, I have managed to improve the quality by using a second-order sigma-delta DAC, but it still has crackling noise audible that shouldn't be there. Has anybody else experimented with sigma-delta DACs and 8-bit audio? If so, what kind quality did you get? I'm wondering if there's something wrong with the DAC, or if I should be doing something like digitally low-pass filtering the 8-bit @ 11 KHz audio data before passing it on to the DAC. Hans

Hi, I just received my Papilio Pro board. I'm newbie to fpga world, and long time user of arduino platform. I wanted to convert my audio library made for arduino Due to the Papilio : http://groovuino.blogspot.fr/ Obviously, I use ZPuino, and until now it works great. I had to change the timer usage, I use the DELTASIGMA (great sound !) instead of Due DAC, had to change some constant definitions, but nothing too difficult (nice job for ZPuino developpers !) Now, of course, I would like to use more advanced features of the Papilio. Here are some questions : . RAM : when I declare a variable, or a table, is it automaticly stored in the RAM ? My goal is to make some audio effects like delays that need a lot of RAM. . FFT : If I understand the FPGA, it's a lot of gates that can be phisicaly programmed to make the functions I need. So I guess it's possible to "program" some FFT functions. I think that ZPuino doesn't use all the FPGA gates (at least on the Papiio Pro), so it must stay some "place" to add these functions. But can it cohabit with ZPuino ? in other words, can I load several systems on my board and make them exchange data ? will I have to learn VHDL ? The FFT would allow me to make some audio effects which need a lot of CPU, like filters, choruses... Thanks in advance, this board seems really great ! Gaétan

To use the Retrocade in a more flexible way, it´s would be really useful, to be able to route the voices to a dedicated single Audio channel. e.g.: SID-Voice 1 to the left Audio-Out Jack 1 SID-Voice 2 to the right Audio-Out Jack 1 SID-Voice 3 to the left Audio-Out Jack 2 YM-Voice 1 to the right Audio-Out Jack 2 (The reason for my question is: I wanna route any single voice to an special external filter for more flexibility) Is it possible in any way? Thx in Advance andY

I just received my RetroCade Synth and it is working great so far But I wanted to ask how the audio output is routed and if it can be changed? Apparantly all output is routed to both audio jacks? Is it planned to allow custom output routing? Also, I assume using one jack as an audio *input* is not possible? Anyway, great job so far! Cheers, Toby